X-Git-Url: http://git.maemo.org/git/?p=mafwsubrenderer;a=blobdiff_plain;f=gst-plugins-base-subtitles0.10%2Fext%2Falsa%2Fgstalsasink.c;fp=gst-plugins-base-subtitles0.10%2Fext%2Falsa%2Fgstalsasink.c;h=2fb37df2a354f6ee8667851d5110e7cc0800cbc5;hp=0000000000000000000000000000000000000000;hb=57ba96e291a055f69dbfd4ae9f1ae2390e36986e;hpb=be2c98fb83895d10ac44af7b9a9c3e00ca54bf49 diff --git a/gst-plugins-base-subtitles0.10/ext/alsa/gstalsasink.c b/gst-plugins-base-subtitles0.10/ext/alsa/gstalsasink.c new file mode 100644 index 0000000..2fb37df --- /dev/null +++ b/gst-plugins-base-subtitles0.10/ext/alsa/gstalsasink.c @@ -0,0 +1,954 @@ +/* GStreamer + * Copyright (C) 2005 Wim Taymans + * Copyright (C) 2006 Tim-Philipp Müller + * + * gstalsasink.c: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-alsasink + * @see_also: alsasrc, alsamixer + * + * This element renders raw audio samples using the ALSA api. + * + * + * Example pipelines + * |[ + * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink + * ]| Play an Ogg/Vorbis file. + * + * + * Last reviewed on 2006-03-01 (0.10.4) + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include +#include +#include +#include +#include +#include +#include + +#include "gstalsa.h" +#include "gstalsasink.h" +#include "gstalsadeviceprobe.h" + +#include + +#define DEFAULT_DEVICE "default" +#define DEFAULT_DEVICE_NAME "" +#define DEFAULT_CARD_NAME "" +#define SPDIF_PERIOD_SIZE 1536 +#define SPDIF_BUFFER_SIZE 15360 + +enum +{ + PROP_0, + PROP_DEVICE, + PROP_DEVICE_NAME, + PROP_CARD_NAME, + PROP_LAST +}; + +static void gst_alsasink_init_interfaces (GType type); + +GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink, + GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces); + +static void gst_alsasink_finalise (GObject * object); +static void gst_alsasink_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_alsasink_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink); + +static gboolean gst_alsasink_open (GstAudioSink * asink); +static gboolean gst_alsasink_prepare (GstAudioSink * asink, + GstRingBufferSpec * spec); +static gboolean gst_alsasink_unprepare (GstAudioSink * asink); +static gboolean gst_alsasink_close (GstAudioSink * asink); +static guint gst_alsasink_write (GstAudioSink * asink, gpointer data, + guint length); +static guint gst_alsasink_delay (GstAudioSink * asink); +static void gst_alsasink_reset (GstAudioSink * asink); + +static gint output_ref; /* 0 */ +static snd_output_t *output; /* NULL */ +static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT; + + +#if (G_BYTE_ORDER == G_LITTLE_ENDIAN) +# define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN" +#else +# define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN" +#endif + +static GstStaticPadTemplate alsasink_sink_factory = + GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 32, " + "depth = (int) 32, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " + "audio/x-raw-int, " + "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 24, " + "depth = (int) 24, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " + "audio/x-raw-int, " + "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 32, " + "depth = (int) 24, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " + "audio/x-raw-int, " + "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 16, " + "depth = (int) 16, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " + "audio/x-raw-int, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 8, " + "depth = (int) 8, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ];" + "audio/x-iec958") + ); + +static void +gst_alsasink_finalise (GObject * object) +{ + GstAlsaSink *sink = GST_ALSA_SINK (object); + + g_free (sink->device); + g_mutex_free (sink->alsa_lock); + + g_static_mutex_lock (&output_mutex); + --output_ref; + if (output_ref == 0) { + snd_output_close (output); + output = NULL; + } + g_static_mutex_unlock (&output_mutex); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_alsasink_init_interfaces (GType type) +{ + gst_alsa_type_add_device_property_probe_interface (type); +} + +static void +gst_alsasink_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_set_details_simple (element_class, + "Audio sink (ALSA)", "Sink/Audio", + "Output to a sound card via ALSA", "Wim Taymans "); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&alsasink_sink_factory)); +} + +static void +gst_alsasink_class_init (GstAlsaSinkClass * klass) +{ + GObjectClass *gobject_class; + GstBaseSinkClass *gstbasesink_class; + GstAudioSinkClass *gstaudiosink_class; + + gobject_class = (GObjectClass *) klass; + gstbasesink_class = (GstBaseSinkClass *) klass; + gstaudiosink_class = (GstAudioSinkClass *) klass; + + parent_class = g_type_class_peek_parent (klass); + + gobject_class->finalize = gst_alsasink_finalise; + gobject_class->get_property = gst_alsasink_get_property; + gobject_class->set_property = gst_alsasink_set_property; + + gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps); + + gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open); + gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare); + gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare); + gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close); + gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write); + gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay); + gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset); + + g_object_class_install_property (gobject_class, PROP_DEVICE, + g_param_spec_string ("device", "Device", + "ALSA device, as defined in an asound configuration file", + DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, + g_param_spec_string ("device-name", "Device name", + "Human-readable name of the sound device", DEFAULT_DEVICE_NAME, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_CARD_NAME, + g_param_spec_string ("card-name", "Card name", + "Human-readable name of the sound card", DEFAULT_CARD_NAME, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); +} + +static void +gst_alsasink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAlsaSink *sink; + + sink = GST_ALSA_SINK (object); + + switch (prop_id) { + case PROP_DEVICE: + g_free (sink->device); + sink->device = g_value_dup_string (value); + /* setting NULL restores the default device */ + if (sink->device == NULL) { + sink->device = g_strdup (DEFAULT_DEVICE); + } + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_alsasink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAlsaSink *sink; + + sink = GST_ALSA_SINK (object); + + switch (prop_id) { + case PROP_DEVICE: + g_value_set_string (value, sink->device); + break; + case PROP_DEVICE_NAME: + g_value_take_string (value, + gst_alsa_find_device_name (GST_OBJECT_CAST (sink), + sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK)); + break; + case PROP_CARD_NAME: + g_value_take_string (value, + gst_alsa_find_card_name (GST_OBJECT_CAST (sink), + sink->device, SND_PCM_STREAM_PLAYBACK)); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class) +{ + GST_DEBUG_OBJECT (alsasink, "initializing alsasink"); + + alsasink->device = g_strdup (DEFAULT_DEVICE); + alsasink->handle = NULL; + alsasink->cached_caps = NULL; + alsasink->alsa_lock = g_mutex_new (); + + g_static_mutex_lock (&output_mutex); + if (output_ref == 0) { + snd_output_stdio_attach (&output, stdout, 0); + ++output_ref; + } + g_static_mutex_unlock (&output_mutex); +} + +#define CHECK(call, error) \ +G_STMT_START { \ +if ((err = call) < 0) \ + goto error; \ +} G_STMT_END; + +static GstCaps * +gst_alsasink_getcaps (GstBaseSink * bsink) +{ + GstElementClass *element_class; + GstPadTemplate *pad_template; + GstAlsaSink *sink = GST_ALSA_SINK (bsink); + GstCaps *caps; + + if (sink->handle == NULL) { + GST_DEBUG_OBJECT (sink, "device not open, using template caps"); + return NULL; /* base class will get template caps for us */ + } + + if (sink->cached_caps) { + GST_LOG_OBJECT (sink, "Returning cached caps"); + return gst_caps_ref (sink->cached_caps); + } + + element_class = GST_ELEMENT_GET_CLASS (sink); + pad_template = gst_element_class_get_pad_template (element_class, "sink"); + g_return_val_if_fail (pad_template != NULL, NULL); + + caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle, + gst_pad_template_get_caps (pad_template)); + + if (caps) { + sink->cached_caps = gst_caps_ref (caps); + } + + GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps); + + return caps; +} + +static int +set_hwparams (GstAlsaSink * alsa) +{ + guint rrate; + gint err; + snd_pcm_hw_params_t *params; + guint period_time, buffer_time; + + snd_pcm_hw_params_malloc (¶ms); + + GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) " + "SPDIF (%d)", alsa->channels, alsa->rate, + snd_pcm_format_name (alsa->format), alsa->iec958); + + /* start with requested values, if we cannot configure alsa for those values, + * we set these values to -1, which will leave the default alsa values */ + buffer_time = alsa->buffer_time; + period_time = alsa->period_time; + +retry: + /* choose all parameters */ + CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config); + /* set the interleaved read/write format */ + CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access), + wrong_access); + /* set the sample format */ + if (alsa->iec958) { + /* Try to use big endian first else fallback to le and swap bytes */ + if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) { + alsa->format = SND_PCM_FORMAT_S16_LE; + alsa->need_swap = TRUE; + GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping"); + } else { + alsa->need_swap = FALSE; + } + } + CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format), + no_sample_format); + /* set the count of channels */ + CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels), + no_channels); + /* set the stream rate */ + rrate = alsa->rate; + CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL), + no_rate); + if (rrate != alsa->rate) + goto rate_match; + +#ifndef GST_DISABLE_GST_DEBUG + /* get and dump some limits */ + { + guint min, max; + + snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL); + snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL); + + GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u", + alsa->buffer_time, min, max); + + snd_pcm_hw_params_get_period_time_min (params, &min, NULL); + snd_pcm_hw_params_get_period_time_max (params, &max, NULL); + + GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u", + alsa->period_time, min, max); + + snd_pcm_hw_params_get_periods_min (params, &min, NULL); + snd_pcm_hw_params_get_periods_max (params, &max, NULL); + + GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max); + } +#endif + + /* now try to configure the buffer time and period time, if one + * of those fail, we fall back to the defaults and emit a warning. */ + if (buffer_time != -1 && !alsa->iec958) { + /* set the buffer time */ + if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params, + &buffer_time, NULL)) < 0) { + GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set buffer time %i for playback: %s", + buffer_time, snd_strerror (err))); + /* disable buffer_time the next round */ + buffer_time = -1; + goto retry; + } + GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time); + } + if (period_time != -1 && !alsa->iec958) { + /* set the period time */ + if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params, + &period_time, NULL)) < 0) { + GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set period time %i for playback: %s", + period_time, snd_strerror (err))); + /* disable period_time the next round */ + period_time = -1; + goto retry; + } + GST_DEBUG_OBJECT (alsa, "period time %u", period_time); + } + + /* Set buffer size and period size manually for SPDIF */ + if (G_UNLIKELY (alsa->iec958)) { + snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE; + snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE; + + CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params, + &buffer_size), buffer_size); + CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params, + &period_size, NULL), period_size); + } + + /* write the parameters to device */ + CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params); + + /* now get the configured values */ + CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size), + buffer_size); + CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL), + period_size); + + GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size, + alsa->period_size); + + snd_pcm_hw_params_free (params); + return 0; + + /* ERRORS */ +no_config: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Broken configuration for playback: no configurations available: %s", + snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +wrong_access: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Access type not available for playback: %s", snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +no_sample_format: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Sample format not available for playback: %s", snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +no_channels: + { + gchar *msg = NULL; + + if ((alsa->channels) == 1) + msg = g_strdup (_("Could not open device for playback in mono mode.")); + if ((alsa->channels) == 2) + msg = g_strdup (_("Could not open device for playback in stereo mode.")); + if ((alsa->channels) > 2) + msg = + g_strdup_printf (_ + ("Could not open device for playback in %d-channel mode."), + alsa->channels); + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg), + ("%s", snd_strerror (err))); + g_free (msg); + snd_pcm_hw_params_free (params); + return err; + } +no_rate: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Rate %iHz not available for playback: %s", + alsa->rate, snd_strerror (err))); + return err; + } +rate_match: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err)); + snd_pcm_hw_params_free (params); + return -EINVAL; + } +buffer_size: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to get buffer size for playback: %s", snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +period_size: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to get period size for playback: %s", snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +set_hw_params: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set hw params for playback: %s", snd_strerror (err))); + snd_pcm_hw_params_free (params); + return err; + } +} + +static int +set_swparams (GstAlsaSink * alsa) +{ + int err; + snd_pcm_sw_params_t *params; + + snd_pcm_sw_params_malloc (¶ms); + + /* get the current swparams */ + CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config); + /* start the transfer when the buffer is almost full: */ + /* (buffer_size / avail_min) * avail_min */ + CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params, + (alsa->buffer_size / alsa->period_size) * alsa->period_size), + start_threshold); + + /* allow the transfer when at least period_size samples can be processed */ + CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params, + alsa->period_size), set_avail); + +#if GST_CHECK_ALSA_VERSION(1,0,16) + /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */ +#else + /* align all transfers to 1 sample */ + CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align); +#endif + + /* write the parameters to the playback device */ + CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params); + + snd_pcm_sw_params_free (params); + return 0; + + /* ERRORS */ +no_config: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to determine current swparams for playback: %s", + snd_strerror (err))); + snd_pcm_sw_params_free (params); + return err; + } +start_threshold: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set start threshold mode for playback: %s", + snd_strerror (err))); + snd_pcm_sw_params_free (params); + return err; + } +set_avail: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set avail min for playback: %s", snd_strerror (err))); + snd_pcm_sw_params_free (params); + return err; + } +#if !GST_CHECK_ALSA_VERSION(1,0,16) +set_align: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set transfer align for playback: %s", snd_strerror (err))); + snd_pcm_sw_params_free (params); + return err; + } +#endif +set_sw_params: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Unable to set sw params for playback: %s", snd_strerror (err))); + snd_pcm_sw_params_free (params); + return err; + } +} + +static gboolean +alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec) +{ + /* Initialize our boolean */ + alsa->iec958 = FALSE; + + switch (spec->type) { + case GST_BUFTYPE_LINEAR: + GST_DEBUG_OBJECT (alsa, + "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth, + spec->width, spec->sign, spec->bigend); + + alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width, + spec->sign ? 0 : 1, spec->bigend ? 1 : 0); + break; + case GST_BUFTYPE_FLOAT: + switch (spec->format) { + case GST_FLOAT32_LE: + alsa->format = SND_PCM_FORMAT_FLOAT_LE; + break; + case GST_FLOAT32_BE: + alsa->format = SND_PCM_FORMAT_FLOAT_BE; + break; + case GST_FLOAT64_LE: + alsa->format = SND_PCM_FORMAT_FLOAT64_LE; + break; + case GST_FLOAT64_BE: + alsa->format = SND_PCM_FORMAT_FLOAT64_BE; + break; + default: + goto error; + } + break; + case GST_BUFTYPE_A_LAW: + alsa->format = SND_PCM_FORMAT_A_LAW; + break; + case GST_BUFTYPE_MU_LAW: + alsa->format = SND_PCM_FORMAT_MU_LAW; + break; + case GST_BUFTYPE_IEC958: + alsa->format = SND_PCM_FORMAT_S16_BE; + alsa->iec958 = TRUE; + break; + default: + goto error; + + } + alsa->rate = spec->rate; + alsa->channels = spec->channels; + alsa->buffer_time = spec->buffer_time; + alsa->period_time = spec->latency_time; + alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED; + + return TRUE; + + /* ERRORS */ +error: + { + return FALSE; + } +} + +static gboolean +gst_alsasink_open (GstAudioSink * asink) +{ + GstAlsaSink *alsa; + gint err; + + alsa = GST_ALSA_SINK (asink); + + /* open in non-blocking mode, we'll use snd_pcm_wait() for space to become + * available. */ + CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK), open_error); + GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device); + + return TRUE; + + /* ERRORS */ +open_error: + { + if (err == -EBUSY) { + GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, + (_("Could not open audio device for playback. " + "Device is being used by another application.")), + ("Device '%s' is busy", alsa->device)); + } else { + GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, + (_("Could not open audio device for playback.")), + ("Playback open error on device '%s': %s", alsa->device, + snd_strerror (err))); + } + return FALSE; + } +} + +static gboolean +gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) +{ + GstAlsaSink *alsa; + gint err; + + alsa = GST_ALSA_SINK (asink); + + if (spec->format == GST_IEC958) { + snd_pcm_close (alsa->handle); + alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa)); + if (G_UNLIKELY (!alsa->handle)) { + goto no_iec958; + } + } + + if (!alsasink_parse_spec (alsa, spec)) + goto spec_parse; + + CHECK (set_hwparams (alsa), hw_params_failed); + CHECK (set_swparams (alsa), sw_params_failed); + + alsa->bytes_per_sample = spec->bytes_per_sample; + spec->segsize = alsa->period_size * spec->bytes_per_sample; + spec->segtotal = alsa->buffer_size / alsa->period_size; + + { + snd_output_t *out_buf = NULL; + char *msg = NULL; + + snd_output_buffer_open (&out_buf); + snd_pcm_dump_hw_setup (alsa->handle, out_buf); + snd_output_buffer_string (out_buf, &msg); + GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg); + snd_output_close (out_buf); + snd_output_buffer_open (&out_buf); + snd_pcm_dump_sw_setup (alsa->handle, out_buf); + snd_output_buffer_string (out_buf, &msg); + GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg); + snd_output_close (out_buf); + } + + return TRUE; + + /* ERRORS */ +no_iec958: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL), + ("Could not open IEC958 (SPDIF) device for playback")); + return FALSE; + } +spec_parse: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Error parsing spec")); + return FALSE; + } +hw_params_failed: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Setting of hwparams failed: %s", snd_strerror (err))); + return FALSE; + } +sw_params_failed: + { + GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), + ("Setting of swparams failed: %s", snd_strerror (err))); + return FALSE; + } +} + +static gboolean +gst_alsasink_unprepare (GstAudioSink * asink) +{ + GstAlsaSink *alsa; + + alsa = GST_ALSA_SINK (asink); + + snd_pcm_drop (alsa->handle); + snd_pcm_hw_free (alsa->handle); + + return TRUE; +} + +static gboolean +gst_alsasink_close (GstAudioSink * asink) +{ + GstAlsaSink *alsa = GST_ALSA_SINK (asink); + + if (alsa->handle) { + snd_pcm_close (alsa->handle); + alsa->handle = NULL; + } + gst_caps_replace (&alsa->cached_caps, NULL); + + return TRUE; +} + + +/* + * Underrun and suspend recovery + */ +static gint +xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err) +{ + GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err); + + if (err == -EPIPE) { /* under-run */ + err = snd_pcm_prepare (handle); + if (err < 0) + GST_WARNING_OBJECT (alsa, + "Can't recovery from underrun, prepare failed: %s", + snd_strerror (err)); + return 0; + } else if (err == -ESTRPIPE) { + while ((err = snd_pcm_resume (handle)) == -EAGAIN) + g_usleep (100); /* wait until the suspend flag is released */ + + if (err < 0) { + err = snd_pcm_prepare (handle); + if (err < 0) + GST_WARNING_OBJECT (alsa, + "Can't recovery from suspend, prepare failed: %s", + snd_strerror (err)); + } + return 0; + } + return err; +} + +static guint +gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length) +{ + GstAlsaSink *alsa; + gint err; + gint cptr; + gint16 *ptr = data; + + alsa = GST_ALSA_SINK (asink); + + if (alsa->iec958 && alsa->need_swap) { + guint i; + + GST_DEBUG_OBJECT (asink, "swapping bytes"); + for (i = 0; i < length / 2; i++) { + ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]); + } + } + + GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length); + + cptr = length / alsa->bytes_per_sample; + + GST_ALSA_SINK_LOCK (asink); + while (cptr > 0) { + /* start by doing a blocking wait for free space. Set the timeout + * to 4 times the period time */ + err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000)); + if (err < 0) { + GST_DEBUG_OBJECT (asink, "wait error, %d", err); + } else { + err = snd_pcm_writei (alsa->handle, ptr, cptr); + } + + GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr); + if (err < 0) { + GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err)); + if (err == -EAGAIN) { + continue; + } else if (xrun_recovery (alsa, alsa->handle, err) < 0) { + goto write_error; + } + continue; + } + + ptr += snd_pcm_frames_to_bytes (alsa->handle, err); + cptr -= err; + } + GST_ALSA_SINK_UNLOCK (asink); + + return length - (cptr * alsa->bytes_per_sample); + +write_error: + { + GST_ALSA_SINK_UNLOCK (asink); + return length; /* skip one period */ + } +} + +static guint +gst_alsasink_delay (GstAudioSink * asink) +{ + GstAlsaSink *alsa; + snd_pcm_sframes_t delay; + int res; + + alsa = GST_ALSA_SINK (asink); + + res = snd_pcm_delay (alsa->handle, &delay); + if (G_UNLIKELY (res < 0)) { + /* on errors, report 0 delay */ + GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res); + delay = 0; + } + if (G_UNLIKELY (delay < 0)) { + /* make sure we never return a negative delay */ + GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay"); + delay = 0; + } + + return delay; +} + +static void +gst_alsasink_reset (GstAudioSink * asink) +{ + GstAlsaSink *alsa; + gint err; + + alsa = GST_ALSA_SINK (asink); + + GST_ALSA_SINK_LOCK (asink); + GST_DEBUG_OBJECT (alsa, "drop"); + CHECK (snd_pcm_drop (alsa->handle), drop_error); + GST_DEBUG_OBJECT (alsa, "prepare"); + CHECK (snd_pcm_prepare (alsa->handle), prepare_error); + GST_DEBUG_OBJECT (alsa, "reset done"); + GST_ALSA_SINK_UNLOCK (asink); + + return; + + /* ERRORS */ +drop_error: + { + GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s", + snd_strerror (err)); + GST_ALSA_SINK_UNLOCK (asink); + return; + } +prepare_error: + { + GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s", + snd_strerror (err)); + GST_ALSA_SINK_UNLOCK (asink); + return; + } +}